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1126 Commits

Author SHA1 Message Date
Randell Jesup
beda527078 Bug 1025354: fix out-of-sync name array for SIPCC logs r=ehugg 2014-06-16 15:10:05 -04:00
Randell Jesup
d96b743305 Bug 1025343: fix issues with overlong codec names in AudioConduit r=pkerr 2014-06-16 01:00:33 -04:00
Randell Jesup
0bedd46970 Bug 1025106: if someone passes us a bogus videocodec config, say it's 'unknown' r=pkerr 2014-06-16 01:00:25 -04:00
Randell Jesup
95ddaacac2 Bug 1022235: Make the webrtc LoadManager/LoadMonitor a singleton r=bsmedberg,pkerr 2014-06-13 15:50:51 -04:00
Randell Jesup
9d1bc6e5a6 Bug 1024288: Add a button to about:webrtc to turn on/off AEC logging r=jib,smaug,unfocused 2014-06-12 12:21:38 -04:00
Randell Jesup
e3e7209c97 Bug 1024288: Allow aec debug data to be dumped on the fly, with max size r=pkerr 2014-06-12 12:20:10 -04:00
Ed Morley
a5c42af943 Backed out changeset 7b4feb3d3a39 (bug 1024288) for compilation errors; CLOSED TREE 2014-06-12 17:41:12 +01:00
Ed Morley
03812e8f9a Backed out changeset 5d63a1316981 (bug 1024288) 2014-06-12 17:40:44 +01:00
Randell Jesup
0a434412d2 Bug 1024288: Add a button to about:webrtc to turn on/off AEC logging r=jib,smaug,unfocused 2014-06-12 12:21:38 -04:00
Randell Jesup
332ff728b8 Bug 1024288: Allow aec debug data to be dumped on the fly, with max size r=pkerr 2014-06-12 12:20:10 -04:00
Randell Jesup
7aefb73722 Bug 1017332: log WebRTC SDP parse errors due to no \n r=ehugg 2014-06-12 12:03:42 -04:00
Byron Campen [:bwc]
881184c858 Bug 1022776 - Bump max transmit count by 1 and modify unit-tests to compensate. r=ekr 2014-06-09 17:31:44 -07:00
Karl Tomlinson
e58f9c45b1 b=1023697 use MediaStream to convert between stream time and seconds/ticks in MediaPipeline r=roc 
The fake graph needs an implementation of the conversion methods.

The real graph will change to use audio ticks for time in a subsequent patch,
but the fake graph doesn't know about MEDIA_TIME_FRAC_BITS, so that change
can be made now in the fake graph.

--HG--
extra : transplant_source : %22%C4%01Yh%5D%F0%A6%11%40%CD%B5%89%0A%8C%8A%C2%19%5E%CC
 2014-06-12 16:44:58 +12:00
Chris Peterson
ce766e4253 Bug 1023075 - Fix more clang warnings in webrtc/signaling. r=jesup 2014-06-09 22:42:11 -07:00
Randell Jesup
3eda6a0803 Bug 970713: Adjust webrtc trace buffering for about:webrtc changes r=pkerr 2014-06-09 04:34:37 -04:00
Jan-Ivar Bruaroey
8b459224fd Bug 970713 - Add 'Start Debug Mode' button to about:webrtc. r=smaug, r=Unfocused, r=jesup 2014-06-08 21:00:12 -04:00
Paul Kerr [:pkerr]
af0b5dd5d3 Bug 970713 - Part 1: Control webrtc logging from about:config settings r=jesup 2014-06-08 18:54:47 -07:00
Randell Jesup
370f28d765 Bug 999704: Implement GMP codec interface to webrtc (not enabled yet) r=joshmoz,ehugg,jesup,pkerr 2014-06-08 17:25:18 -04:00
Ryan VanderMeulen
0ae54304d5 Backed out changeset 2af237fa2079 (bug 999704) for bustage. 
CLOSED TREE DONTBUILD
 2014-06-08 14:39:44 -04:00
Randell Jesup
8cf755ddd9 Bug 999704: Implement GMP codec interface to webrtc (not enabled yet) r=joshmoz,ehugg,jesup 2014-06-08 14:07:53 -04:00
Randell Jesup
442154b7cb Bug 970742: Add receive state monitoring to webrtc CodecStatistics r=jib 2014-06-08 11:06:30 -04:00
Randell Jesup
fc5f6c61d2 Bug 970742: Monitor decoder error state to enable recording errors and error recovery times r=jib 2014-06-08 10:33:02 -04:00
Jan-Ivar Bruaroey
73c28df208 Bug 951496 - Codec telemetry. r=jesup 2014-06-07 17:33:39 -04:00
Jan-Ivar Bruaroey
f23107dd2f Bug 951496 - Codec getStats. r=smaug, r=jesup 2014-06-07 17:27:26 -04:00
Steven Lee
96d69b8623 Bug 951496 - Statistics data for checking the status of codec. r=jesup 2014-06-04 23:56:30 -04:00
Jan-Ivar Bruaroey
12dfa6e7da Bug 951496 - Fix Stastistics typo in vie_codec. r=jesup 2014-06-04 23:57:02 -04:00
Adam Roach [:abr]
df82c8e1e7 Bug 1018372 - Check aThread against mThread in PeerConnectionImpl constructor r=jesup 2014-06-06 15:56:47 -05:00
Karl Tomlinson
0b9ed65c05 b=1015828 match Fake_MediaStreamListener::NotifyPull time advances to timer period and Fake_AudioStreamSource::Periodic buffer size r=rjesup 
Also, increment Fake_SourceMediaStream::mDesiredTime each period,
instead of each listener notification.

--HG--
extra : rebase_source : 723a2a3b126adca486154d0b686746c21dbac37e
 2014-06-05 10:11:51 +12:00
Randell Jesup
cd089192fd Bug 1003712: Codec availability support and prioritization r=ehugg 2014-06-04 14:52:32 -04:00
Randell Jesup
7e84082c49 Bug 1003712: Use Media Resource Manager to reserve OMX Codecs r=jhlin 2014-06-04 14:52:31 -04:00
Byron Campen [:bwc]
bbaf4386c7 Bug 998989 - Part 1: ChromeOnly API for getting notifications when PCs are initted, or change ICE connection/gathering state. Also, expose the PC id, and allow getAllStats to be filtered by the same. r=jib, r=bz 2014-05-22 14:14:56 -07:00
Robert O'Callahan
a8bbe633b9 Bug 1015664. Part 2: Remove some NS_HIDDEN usage. r=bsmedberg 2014-06-03 00:08:24 +12:00
EKR
4884fdde56 Bug 1018473. Unit test for duplicate trickle candidates. r=bwc 2014-05-31 12:06:45 -07:00
Byron Campen [:bwc]
7b0fa364cc Bug 1018473: Detect when vcmRxAllocICE has already been called for a given stream, and suppress the (duplicate) connection to SignalCandidate. r=ekr 2014-05-31 19:41:53 -07:00
Byron Campen [:bwc]
1774026f94 Bug 1017291 - Keep track of the number of errors in AddIceCandidate before ICE completes, and record this number in telemetry in the success and failure cases separately. r=ekr 2014-05-29 08:40:31 -07:00
Mike Hommey
41657ceb81 Fix non-unified build bustage from bug 987979 on a CLOSED TREE. r=me 2014-05-30 09:32:08 +09:00
Randell Jesup
5aaae2b64e Bug 987979: Patch 12 - Add webrtc JNI target annotations to stop ProGuard from removing too much code. r=blassey 2014-05-29 17:05:16 -04:00
Randell Jesup
d65b42fede Bug 987979: Patch 11 - Add webrtc 3.50 support for Froyo/Gingerbread/Ice Cream Sandwich. r=blassey 2014-05-29 17:05:16 -04:00
Randell Jesup
5b9598c2f6 Bug 987979: Patch 10 - Support building with older Android SDKs. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
8cb8e97704 Bug 987979: Patch 9 - Use Android JNI Wrappers for off-thread FindClass and Global Context. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
12d756308d Bug 987979: Patch 8 - Support rotating/suspending/resuming an ongoing WebRTC call. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
a8d21229d7 Bug 987979: Patch 7 - Remove JSON/UCI requirements for Camera capture capability. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
e2805a0c2d Bug 987979: Patch 6 - Include CPU feature detection source directly. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
500b3d6ff7 Bug 987979: Patch 5 - Enable switching between OpenSLES and JNI backends, dummy OpenSLES output. r=rjesup 2014-05-29 17:05:14 -04:00
Randell Jesup
4465789496 Bug 987979: Patch 4 - Rework WebRTC.org audio code for Mozilla integration. r=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
79df25773b Bug 987979: Patch 3 - Fix various build issues in webrtc.org/Mozilla integration. r=rjesup 2014-05-29 17:05:14 -04:00
Randell Jesup
7740e2ceb2 Bug 987979: Patch 2 - Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
66465cce72 Bug 987979: Patch 1 - Webrtc updated to branch 3.50 rev 5764, pull made Mon Mar 24 15:36:34 EDT 2014 rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/video_engine/new_include/config.h => media/webrtc/trunk/webrtc/config.h
rename : media/webrtc/trunk/webrtc/video_engine/new_include/frame_callback.h => media/webrtc/trunk/webrtc/frame_callback.h
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java => media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
rename : media/webrtc/trunk/webrtc/common_unittest.cc => media/webrtc/trunk/webrtc/test/common_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/direct_transport.h => media/webrtc/trunk/webrtc/test/direct_transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_decoder.cc => media/webrtc/trunk/webrtc/test/fake_decoder.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_decoder.h => media/webrtc/trunk/webrtc/test/fake_decoder.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_encoder.cc => media/webrtc/trunk/webrtc/test/fake_encoder.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_encoder.h => media/webrtc/trunk/webrtc/test/fake_encoder.h
rename : media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe_unittest.cc => media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/flags.cc => media/webrtc/trunk/webrtc/test/flags.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/flags.h => media/webrtc/trunk/webrtc/test/flags.h
rename : media/webrtc/trunk/webrtc/common_video/test/frame_generator.h => media/webrtc/trunk/webrtc/test/frame_generator.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/frame_generator_capturer.cc => media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/frame_generator_capturer.h => media/webrtc/trunk/webrtc/test/frame_generator_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/gl/gl_renderer.cc => media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/gl/gl_renderer.h => media/webrtc/trunk/webrtc/test/gl/gl_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/glx_renderer.cc => media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/glx_renderer.h => media/webrtc/trunk/webrtc/test/linux/glx_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/video_renderer_linux.cc => media/webrtc/trunk/webrtc/test/linux/video_renderer_linux.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/run_tests.mm => media/webrtc/trunk/webrtc/test/mac/run_tests.mm
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/video_renderer_mac.h => media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/video_renderer_mac.mm => media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_platform_renderer.cc => media/webrtc/trunk/webrtc/test/null_platform_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_transport.cc => media/webrtc/trunk/webrtc/test/null_transport.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_transport.h => media/webrtc/trunk/webrtc/test/null_transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/rtp_rtcp_observer.h => media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/run_loop.h => media/webrtc/trunk/webrtc/test/run_loop.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/run_tests.h => media/webrtc/trunk/webrtc/test/run_tests.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/statistics.cc => media/webrtc/trunk/webrtc/test/statistics.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/statistics.h => media/webrtc/trunk/webrtc/test/statistics.h
rename : media/webrtc/trunk/webrtc/video_engine/test/test_main.cc => media/webrtc/trunk/webrtc/test/test_main.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/vcm_capturer.cc => media/webrtc/trunk/webrtc/test/vcm_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/vcm_capturer.h => media/webrtc/trunk/webrtc/test/vcm_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_capturer.cc => media/webrtc/trunk/webrtc/test/video_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_capturer.h => media/webrtc/trunk/webrtc/test/video_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_renderer.cc => media/webrtc/trunk/webrtc/test/video_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_renderer.h => media/webrtc/trunk/webrtc/test/video_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/d3d_renderer.cc => media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/d3d_renderer.h => media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/run_loop_win.cc => media/webrtc/trunk/webrtc/test/win/run_loop_win.cc
rename : media/webrtc/trunk/webrtc/video_engine/new_include/transport.h => media/webrtc/trunk/webrtc/transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/loopback.cc => media/webrtc/trunk/webrtc/video/loopback.cc
rename : media/webrtc/trunk/webrtc/video_engine/internal/transport_adapter.cc => media/webrtc/trunk/webrtc/video/transport_adapter.cc
rename : media/webrtc/trunk/webrtc/video_engine/internal/transport_adapter.h => media/webrtc/trunk/webrtc/video/transport_adapter.h
rename : media/webrtc/trunk/webrtc/video_engine/new_include/video_renderer.h => media/webrtc/trunk/webrtc/video_renderer.h
 2014-05-29 17:05:13 -04:00
Nils Ohlmeier [:drno]
5abd2eec9b Bug 1014304 - Remove onconnection and onclosedconnection from RTCPeerConnection. r=jib, r=jesup, r=smaug 2014-05-28 09:36:00 -04:00
Jan-Ivar Bruaroey
d30b322032 Bug 859565 - Remove legacy PeerConnectionImpl.readyState. r=bholley, r=abr 2014-05-17 17:11:27 -04:00
Byron Campen [:bwc]
caa58710f6 Bug 1016724 - Make sure the word "gathering" appears in the timecard stamp for complete. r=jesup 2014-05-27 17:19:45 -07:00
Randell Jesup
51036cd19e Bug 743703: allow mirroring of trace logs to NSPR; fix backwards lazy allocation defines r=pkerr 2014-05-28 03:18:33 -04:00
Enda Mannion
2a2092dc76 Bug 1003994 - H.246 and multiple video codec tests. r=jesup 2014-05-26 10:07:19 +01:00
Jan-Ivar Bruaroey
93980709ca Bug 970685, telemetry for WebRTC bandwidth, stats-tweak approach. r=jesup 2014-05-27 14:41:17 -04:00
Jan-Ivar Bruaroey
723947759f Bug 970685 - tweak internal RTCStatsQuery to use nsAutoPtr for report, so it can be stolen 2014-05-27 12:58:03 -04:00
EKR
df92fcf432 Bug 1015409 - Fix trickle between CreateOffer() and SetLocal(). r=bwc 2014-05-27 13:13:43 -07:00
Randell Jesup
3e5d95d980 Bug 1014819: Replace OMX GetCodecConfig with straight caching of H.264 SPS/PPS r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
c102072fa1 Bug 985254: Modify H264 OMX code to deal with upstream code inserting start codes r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
388a6314ad Bug 1014921: Wallpaper 8x10 OMX H264 encode/decode mismatch by forcing IDRs r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
ff9cdaf529 Bug 997567: Send iframes for HW H264 encoder when bitrate changes with long GOP r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
d8de28f2bb Bug 997567: Support reconfiguration for frame-rate changes on OMX H.264 encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
bc26a0f3d5 Bug 1015029: Use OMX_VIDEO_ControlRateConstantSkipFrames mode for H.264 OMX encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
6143419992 Bug 989945: add a bit more logging to H264 OMX codec r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
9216b39c92 Bug 989945: Use configureDirect to set OMX HW H264 encoder config correctly r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
06ca7ee4bf Bug 989945: increase logging for H264 OMX code r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
88dd15fc1e Bug 985253: Support H.264 RTP mode 1 support in webrtc signaling r=ehugg 2014-05-24 18:28:02 -04:00
Randell Jesup
41ccb95961 Bug 985254: Add H264 codec-specific structure to carry negotiated data r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
fd032ddd4c Bug 985254: Cleaup H264 packetization and jitter buffer r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
295343b36e Bug 985254: modify upstream h264 packetization patch to make it work r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
fe6e9b77c7 Bug 985254: review cleanups from H264 packetization patch r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
72962eb159 Bug 985254: H264 RTP packetization imported from upstream patchset #10 r=jesup 
https://webrtc-codereview.appspot.com/13399004/
 2014-05-24 18:28:00 -04:00
Randell Jesup
3683c22e00 Bug 1004396: Make video codec default bitrates configurable for WebRTC r=ekr 2014-05-24 18:28:00 -04:00
Kyle Huey
8a1ded0d50 Bug 996133: Remove unnecessary NS_DISPATCH_NORMAL arguments to NS_DispatchToMainThread. r=ehsan 2014-05-23 12:53:17 -07:00
Jan-Ivar Bruaroey
a652a7f255 Bug 1013238 - Fix timer event crash on shutdown in recent PeerConnectionCtx change. r=jesup 2014-05-21 22:32:03 -04:00
Anders Lund
4fb3d4f35c Bug 942188 - Added parsing of ice-lite attribute and start ice checks as controlling if peer is ice-lite. r=abr 2014-05-16 01:32:00 -05:00
Byron Campen [:bwc]
a06fbf3117 Bug 1013729 - Null check in case PushLayers failed when registering for the DTLS connection signal. r=jesup 2014-05-21 08:49:03 -07:00
Carsten "Tomcat" Book
a161c852d2 Backed out changeset 9b2588d41e3a (bug 969395) for bustage 2014-05-21 11:29:21 +02:00
Qiang Lu
0869002d91 Bug 969395 - Add stub library for accesing VP8 HW codec through android native mediacodec interface. r=rjesup 2014-05-21 10:14:31 +08:00
EKR
b0ca419e1e Bug 1012999: When STUN global rate limit is exceeded, record this in telemetry. r=ekr 2014-05-19 19:16:38 -07:00
Jan-Ivar Bruaroey
904d9cd547 Bug 970685 - Thread approach for WebRTC telemetry for jitter, packet-loss and RTT. r=jesup 2014-05-10 08:54:41 -04:00
John Lin
335a321bf8 Bug 1011422 - Clear mOMXConfigured flag to correctly restart OMX H.264 encoder. r=jesup 2014-05-18 19:30:00 +02:00
Carsten "Tomcat" Book
c46f2f1ad5 Backed out changeset 426b187eae45 (bug 1001422) wrong bugnumber in commit message 2014-05-19 11:44:00 +02:00
John Lin
1b10c80f8a Bug 1001422 - Clear mOMXConfigured flag to correctly restart OMX H.264 encoder. r=jesup 2014-05-18 19:30:00 +02:00
John Lin
4482fb3908 Bug 1010841 - Handle on-demand key frame request in OMX H.264 encoder. r=jesup 2014-05-16 01:56:00 -04:00
Randell Jesup
9e7930e7fd Bug 1011214: Release OMX monitor when shutting down Encoder output drain thread r=jhlin 2014-05-16 04:37:08 -04:00
Randell Jesup
f83fd207cc Bug 981780: fix disable-webrtc r=glandium 2014-05-09 14:40:32 -04:00
Martin Thomson
e53bb3766c Bug 966066 - Add principal observer to RTCPeerConnection. r=jib 2014-04-25 10:34:00 -04:00
Neil Rashbrook
0b29793db8 Bug 514280 Only use nsCOMPtr for interfaces r=bsmedberg 2014-05-11 10:47:11 +01:00
Ryan VanderMeulen
d30bf9e6eb Backed out changeset 047f98eef5cf (bug 1007196) for intermittent failures. 2014-05-09 14:13:21 -04:00
Ethan Hugg
273eb13fb2 Bug 1007196 - Re-enable the Signaling unittests for Linux and OSX. r=ted 2014-05-07 13:04:34 -07:00
Chris Peterson
df47d0f97f Bug 990764 - Replace MOZ_ASSUME_UNREACHABLE in webrtc/signaling. r=jesup 2014-04-19 11:05:10 -07:00
Neil Rashbrook
a998ae77f6 Backout of bug 514280 changeset c738f7348dea for build failure on a CLOSED TREE 2014-05-08 20:35:09 +01:00
Neil Rashbrook
f9520ae677 Bug 514280 Only use nsCOMPtr for interfaces r=bsmedberg 2014-05-08 20:08:38 +01:00
Chris Peterson
78ae1f032d Bug 1005784 - Fix -Wuninitialized warnings in webrtc/modules/audio_device/linux/. r=jesup 2014-05-05 23:38:04 -07:00
Byron Campen [:bwc]
ed80aecb7c Bug 1002831 - Display remote and local SDP on about:webrtc. r=smaug, r=jib 2014-05-05 11:13:24 -07:00
Byron Campen [:bwc]
a0a82f96e2 Bug 970734 - Part 2: Record final ICE/media stats when PeerConnections are closed, so they show up in about:webrtc. r=smaug, r=jib 2014-05-05 09:35:57 -07:00
Robert O'Callahan
730a55616d Bug 1006248. Part 4: Use better #include paths for libstagefright headers in a couple of places. r=glandium 
--HG--
extra : rebase_source : e8c7e019b0bc5bf60081aad158a7d89fbb261e29
 2014-05-06 17:40:59 +12:00
Martin Thomson
33feea0ff7 Bug 1006112 - Fixing regressions in signaling_unittests. r=ekr 2014-05-05 14:19:00 +02:00
Martin Thomson
8b16c70404 Bug 942367 - Stream isolation for WebRTC r=bholley 2014-05-01 12:51:00 +02:00
Ethan Hugg
2da17470dd Bug 1002890 - Signaling unittests no longer need exceptions to mainthread checks. r=jesup 2014-04-28 19:45:40 -07:00
Ethan Hugg
4262180c0b Bug 819549 - Signaling unittests should dispatch to main thread when calling PC. r=jesup 2014-04-28 15:00:19 -07:00
Randell Jesup
1a11d079ec Bug 985253: Send rtcp-fb for all video codecs, and fix answer generation for H.264 for rtcp-fb r=ehugg 2014-04-30 18:18:35 -04:00
John Lin
80b73a43cf Bug 1002402: typo fix for adjusting SPS/PPS timestamps r=jesup 2014-04-30 01:20:41 -04:00
John Lin
8e3ce5bc7e Bug 1002402: (temporary) change SPS/PPS timestamps so webrtc jitter buffer won't drop them r=jesup 2014-04-29 13:25:40 -04:00
Ed Morley
42a7f8cefd Merge mozilla-central and inbound 2014-04-29 18:23:29 +01:00
Randell Jesup
0a0df84f20 Bug 1002306: don't accidentally disable non-H264 codecs in the OMX code r=ekr 2014-04-28 19:52:16 -04:00
John Lin
93944b130b Bug 911046 - Get graphic buffers of decoded frames through gonk native window callback. r=jesup 2014-04-27 21:07:00 -04:00
John Lin
28c564f273 Bug 1002402 - Support RTP H.264 input data in WebRTC OMX decoder. r=jesup 2014-04-28 23:37:00 +02:00
Byron Campen [:bwc]
035bbcada2 Bug 1001959 - Give up references to NrIceMediaStream on STS instead of main. r=jib 2014-04-28 09:01:29 -07:00
Birunthan Mohanathas
ff8ce9bd42 Bug 900908 - Part 3: Change uses of numbered macros in nsIClassInfoImpl.h/nsISupportsImpl.h to the variadic variants. r=froydnj 2014-04-27 03:06:00 -04:00
Garvan Keeley
5d9d95fc9b Bug 1001708: Don't use ternary operator with class const statics r=jesup 2014-04-27 00:02:17 -04:00
Byron Campen [:bwc]
82e3ffd1c2 Bug 970690 - Part 2: Add basic telemetry for ICE. r=mt 2014-03-03 10:58:35 -08:00
Martin Thomson
8e0eab87a9 Bug 1001539 - Fix compilation warning in ccsip_pmh.c. r=bwc 2014-04-25 10:58:00 -04:00
Paul Kerr [:pkerr]
4c183baabe Bug 970691 - Part 2: Restore digit stamping function to YuvStamper. r=jesup 
Refactor digit writing method to use the new internals. Allows digit string
to wrap through multiple lines in a small frame.
 2014-04-24 19:58:21 -07:00
Paul Kerr [:pkerr]
2435cef6ad Bug 970691 - Part 1: Add timestamp to fake video. r=jesup 
Update YuvStamper utility. Add a CRC32 to the encoded
payload and have the decode method us this to verify reception.
Wrap encoded values across multiple lines in the frame buffer
when necessary. Use YuvStamper to encode a timestamp in each fake video frame.
Extract the value in VideoConduit to calculate the video latency
and add this to a running average latency when enabled via config.
 2014-03-22 16:35:43 -07:00
John
aec1baf311 Bug 999902 - Enable WebRTC OMX codec only when Android version >= 18. r=jesup 2014-04-23 02:59:00 +02:00
Wes Kocher
bd5e387b25 Backed out 2 changesets (bug 970691) for build bustage on a CLOSED TREE 
Backed out changeset 83f7aec5a083 (bug 970691)
Backed out changeset 94348d189ed5 (bug 970691)
 2014-04-23 18:26:05 -07:00
Paul Kerr [:pkerr]
81d99db985 Bug 970691 - Part2: Restore digit stamping function to YuvStamper. r=jesup 
Refactor digit writing method to use the new internals. Allows digit string
to wrap through multiple lines in a small frame.
 2014-04-23 10:03:18 -07:00
Paul Kerr [:pkerr]
8a77293ef3 Bug 970691 - Part 1: Add timestamp to fake video. r=jesup 
Update YuvStamper utility. Add a CRC32 to the encoded
payload and have the decode method us this to verify reception.
Wrap encoded values across multiple lines in the frame buffer
when necessary. Use YuvStamper to encode a timestamp in each fake video frame.
Extract the value in VideoConduit to calculate the video latency
and add this to a running average latency when enabled via config.
 2014-03-22 16:35:43 -07:00
John Lin
c6c26a7758 Bug 911046 - Part 6: Make H.264 preferred video codec when enabled in preferences. r=jesup, ekr 2014-04-21 23:44:00 +02:00
John Lin
80b58803b4 Bug 911046 - Part 5: Register H.264 external codec for B2G. r=jesup, ekr 2014-04-21 23:43:00 +02:00
John Lin
51b3bae2a3 Bug 911046 - Part 4: Add external H.264 HW video codec implementation for B2G. r=jesup 2014-04-21 23:42:00 +02:00
John Lin
9dcd47e846 Bug 911046 - Part 2: Support 'handle-using' video frames for WebRTC on B2G. r=jesup, ekr 2014-04-21 23:41:00 +02:00
John Lin
50cc31714b Bug 911046 - Part 1: Support external codec in VideoConduit. r=jesup 2014-04-21 23:40:00 +02:00
Ethan Hugg
ac0270a5ba Bug 995380 - Signaling unittests should use the real main thread. r=jesup 2014-04-21 19:37:22 -07:00
Ryan VanderMeulen
a9cc5ff586 Backed out changesets 1e581e74878d, 7d2138e87ca0, and 7cc66aee4341 (bug 942367) for B2G mochitest failures. 
CLOSED TREE
 2014-04-17 22:26:07 -04:00
Randell Jesup
061c1534da Bug 996853: handle AUDIO_FORMAT_SILENCE in MediaPipeline and AudioSegment::WriteTo r=roc 2014-04-17 17:45:25 -04:00
Martin Thomson
41016bdb71 Bug 942367 - Part 3: Stream isolation for WebRTC. r=jib, r=bholley 2014-04-10 11:52:08 -07:00
Nils Ohlmeier [:drno]
1ba4742ccf Bug 989936 - fire the onsignalingstatechanged event if close was called locally. r=jesup 2014-04-16 18:02:00 +02:00
Carsten "Tomcat" Book
85d97d12db Backed out changeset e6c72bcaa64c (bug 942367) 2014-04-16 09:54:31 +02:00
Martin Thomson
b068285328 Bug 942367 - Stream isolation for WebRTC. r=jib,bholley 2014-04-15 14:36:00 +02:00
Jonathan Watt
0a0470e24e Bug 996901 - Remove lots of gfxASurface.h and gfxImageSurface.h includes and forward declarations that are no longer needed. r=mattwoodrow 2014-04-16 01:41:40 +01:00
Randell Jesup
f0300f6626 Bug 996329: remove trailing space from m=application SDP lines r=ehugg 2014-04-15 14:00:59 -04:00
Nils Ohlmeier [:drno]
9ee18bdbee Bug 993780 - Ignore calls to SetSignalingState_m once PC is in close. r=jib,rjesup 2014-04-10 14:55:00 +02:00
Nils Ohlmeier [:drno]
85efe0d58e Bug 994999 - Rename IsClosed() to HasMedia() and let IsClosed() return SignalingState instead. r=jesup, r=bwc 2014-04-13 16:17:51 -04:00
Ryan VanderMeulen
411811f032 Merge m-c to inbound on a CLOSED TREE. 2014-04-11 16:24:56 -04:00
Sotaro Ikeda
8305355d18 Bug 990310 - Remove SurfaceDescriptor from media and GrallocImage r=nical,cajbir 2014-04-11 06:13:12 -07:00
Randell Jesup
d63ba60c75 Bug 694814: Patch 5 - Move AEC from PeerConnection to getUserMedia rs=padenot 2014-04-02 13:58:19 -04:00
Randell Jesup
46a6b9385e Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted 2014-04-02 13:58:19 -04:00
Randell Jesup
8fbb6219f2 Bug 694814: Patch 1: Add farend input to webrtc.org upstream rs=padenot 2014-04-02 13:58:19 -04:00
Paul Adenot
4c111084f0 Bug 818822 - Update AudioConduit so it can work at 44.1kHz. r=jesup 2014-03-24 11:06:05 +01:00
Byron Campen [:bwc]
b4264cd392 Bug 993141 - Fix bug where we were assuming DataChannel::mStream corresponded to the level. r=jib 2014-04-07 15:21:06 -07:00
Boris Zbarsky
35fca5eeeb Bug 991742 part 8. Remove the "aScope" argument of WebIDL/nsWrapperCache WrapObject() methods. r=bholley 
This patch was mostly generated with the following command:

find . -name "*.h" -o -name "*.cpp" | xargs sed -e '/WrapObject(JSContext/ {; N; s/\(WrapObject(JSContext *\* *a\{0,1\}[Cc]x\),\n\{0,1\} *JS::Handle<JSObject\*> a\{0,1\}[sS]cope/\1/ ; }' -i ""

and then reverting the changes that made to
dom/bindings/BindingUtils.h, since those WrapObject methods are not
the ones we're trying to change here, plus a bunch of manual fixups
for cases that this command did not catch (including all the callsites
of WrapObject()).
 2014-04-08 18:27:18 -04:00
Boris Zbarsky
56f44fdf10 Bug 991742 part 6. Remove the "aScope" argument of binding Wrap() methods. r=bholley 
This patch was mostly generated with this command:

find . -name "*.h" -o -name "*.cpp" | xargs sed -e 's/Binding::Wrap(aCx, aScope, this/Binding::Wrap(aCx, this/' -e 's/Binding_workers::Wrap(aCx, aScope, this/Binding_workers::Wrap(aCx, this/' -e 's/Binding::Wrap(cx, scope, this/Binding::Wrap(cx, this/' -i ""

plus a few manual fixes to dom/bindings/Codegen.py, js/xpconnect/src/event_impl_gen.py, and a few C++ files that were not caught in the search-and-replace above.
 2014-04-08 18:27:17 -04:00
Peter Van der Beken
c0b23e34f5 Bug 990158 - Make inner windows use their wrapper cache. r=bz. 
--HG--
extra : rebase_source : bc040c75280bb45ae7ab0ed302130ff5d7178153
 2013-11-09 11:20:22 +01:00
Randell Jesup
c47531536e Backed out changeset 33072f5b4c66 (bug 818822) 2014-04-07 15:37:57 -04:00
Randell Jesup
2ade2a2cdc Backed out changeset 89a615263614 (bug 694814) 2014-04-07 15:37:55 -04:00
Randell Jesup
373d268aa8 Backed out changeset 6922b1261595 (bug 694814) 2014-04-07 15:37:54 -04:00
Randell Jesup
c824757a77 Backed out changeset 6dc08e9fc7e8 (bug 694814) 2014-04-07 15:37:50 -04:00
Randell Jesup
9f3e338198 Bug 694814: Patch 5 - Move AEC from PeerConnection to getUserMedia rs=padenot 2014-04-02 13:58:19 -04:00
Randell Jesup
29ba637c69 Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted 2014-04-02 13:58:19 -04:00
Randell Jesup
73e3825d95 Bug 694814: Patch 1: Add farend input to webrtc.org upstream rs=padenot 2014-04-02 13:58:19 -04:00
Paul Adenot
dcabfd1773 Bug 818822 - Update AudioConduit so it can work at 44.1kHz. r=jesup 2014-03-24 11:06:05 +01:00
Matt Woodrow
b93f84571b Bug 991028 - Remove deprecated IPDL SurfaceDescriptor types. r=nical 2014-04-07 13:32:49 +12:00
Phil Ringnalda
18f873ac9e Backed out 4 changesets (bug 991028) for nonunified bustage 
CLOSED TREE

Backed out changeset 147581a518c3 (bug 991028)
Backed out changeset e5bacc566e58 (bug 991028)
Backed out changeset 6dc852777a4d (bug 991028)
Backed out changeset 780bec5571b9 (bug 991028)
 2014-04-06 21:21:38 -07:00
Matt Woodrow
0851e5c863 Bug 991028 - Remove deprecated IPDL SurfaceDescriptor types. r=nical 2014-04-07 13:32:49 +12:00
Randell Jesup
beb3941cd7 Backed out 965c62289427:cb894b5d342f for perma-orange on b2g emulator M10 r=backout 2014-04-02 17:11:12 -04:00
Randell Jesup
c79d51ae9c Bug 694814: Patch 5 - Move AEC from PeerConnection to getUserMedia rs=padenot 2014-04-02 13:58:19 -04:00
Randell Jesup
b3a497d253 Bug 694814: Patch 2: modifications to webrtc.org single_rw_fifo r=glandium,ted 2014-04-02 13:58:19 -04:00
Randell Jesup
40fe624598 Bug 694814: Patch 1: Add farend input to webrtc.org upstream rs=padenot 2014-04-02 13:58:19 -04:00
Paul Adenot
af2f4c481b Bug 818822 - Update AudioConduit so it can work at 44.1kHz. r=jesup 2014-03-24 11:06:05 +01:00
snigdha
90f0f64d5b Bug 798033 - Headers should generally not do "using namespace" at file scope. r=jib, r=jmathies, r=rjesup, r=ekr, r=ncameron, r=blassey 2014-04-01 08:29:25 -04:00
Daniel Holbert
a00a50512c Bug 989425: Remove unused variable 'DTLS_FINGERPRINT_LENGTH' from PeerConnectionImpl.cpp. r=mt 2014-03-28 17:58:19 -07:00
Randell Jesup
67919b2f71 Bug 986764: release all webrtc sub-modules before deleting engine r=gcp 2014-03-26 17:58:25 -04:00
Jan Beich
a0cc624457 Bug 985848 - Use videodev2.h on DragonFly/DPorts as well. r=jesup 2014-03-24 08:57:58 -04:00
Jan-Ivar Bruaroey
cd8307646e Bug 964127 - Add a/v sync telemetry. r=bwc 2014-03-14 16:46:31 -04:00
Kyle Huey
84360900b0 Bug 967364: Pass already_AddRefed by reference instead of by value. r=bsmedberg 2014-03-15 12:00:17 -07:00
Kyle Huey
510a49016d Bug 967364: Rename already_AddRefed::get to take. r=bsmedberg 2014-03-15 12:00:15 -07:00
Jan-Ivar Bruaroey
b31835c861 Bug 970686: Undo extensions to wrong rtcp methods in webrtc.org r=jesup 2014-03-13 22:28:12 -04:00
Jan-Ivar Bruaroey
57b144b21b Bug 970686: Outbound getStats: Fixed RTCP timestamps and remote packets/bytes received. r=jesup 2014-03-14 14:34:02 -04:00
Ethan Hugg
1776571e5f Bug 982371 - Signaling - Filter remotepartyname in the calllogger. r=jesup 2014-03-12 12:04:18 -07:00
Gian-Carlo Pascutto
5eb3621aee Bug 877954 - Add additional logging for WebRTC adaption & resolution changes. r=jesup 2014-03-13 11:06:39 +01:00
Gian-Carlo Pascutto
739c10fdfb Bug 877954 - Enable QM if load adaption is enabled. r=jesup 2014-03-13 11:06:27 +01:00
Gian-Carlo Pascutto
fece01ade1 Bug 877954 - Push load state to media optimization. Add simple CPU adaption rules. r=jesup 2014-03-13 11:05:42 +01:00
Gian-Carlo Pascutto
3fefb76168 Bug 877954 - Implement Load Management service. Add callbacks to ViEncoder. r=jesup 2014-03-13 11:05:27 +01:00
Jan-Ivar Bruaroey
fe2bc5a3f5 Bug 964127: Add a/v sync status to about:webrtc. r=jesup 2014-03-12 17:13:20 -04:00
Randell Jesup
da2a501066 Bug 964127: Add logging of webrtc a/v sync status r=jib 2014-03-12 20:11:49 -04:00
Ehsan Akhgari
98311f51be Bug 981428 - Move OSX -framework flags to moz.build; r=mshal 2014-03-10 20:18:33 -04:00
Randell Jesup
3c46e23828 Bug 981680: Upstream webrtc patch for avsync (r5102) rs=jesup 2014-03-11 00:36:12 -04:00
Byron Campen [:bwc]
1391072dab Bug 979471 - Populate ICE stats for DataChannels when we have a null selector. r=jib 2014-03-10 08:41:35 -07:00
Randell Jesup
4b980a852d Bug 980096: fix leaks of VoiceEngines by reinstating use of ScopedCustomReleasePtr r=khuey 
--HG--
rename : media/webrtc/signaling/src/media-conduit/MediaEngineWrapper.h => media/webrtc/signaling/src/common/MediaEngineWrapper.h
 2014-03-09 00:18:50 -05:00
Byron Campen [:bwc]
1e9703c504 Bug 786234 - Part 6: Move RTCP handling back to the transmit pipeline. r=jesup, r=ethanhugg 2014-02-25 09:22:31 -08:00
Byron Campen [:bwc]
cd1137e11e Bug 786234 - Part 5: More detailed test-cases. r=ehugg 2014-01-17 17:11:00 -08:00
Byron Campen [:bwc]
a3eb800b88 Bug 786234 - Part 4: Fix a bug in mediapipeline_unittest where RTP packets were 
mashed together into a single buffer five at a time, preventing the receive
pipeline from successfully decrypting them, and causing the packet counts to
be wrong. (This should fix 947663) r=ehugg
 2014-01-17 17:10:17 -08:00
Byron Campen [:bwc]
964dc99bb7 Bug 786234 - Part 3: Plumbing for filtration, and bundle usage detection. r=abr 2014-01-14 16:29:42 -08:00
Byron Campen [:bwc]
e7761b04eb Bug 786234 - Part 2.2: Compensate for some build system gotchas. r=abr 2014-01-14 16:31:27 -08:00
Byron Campen [:bwc]
83869a10c0 Bug 786234 - Part 2.1: RTCP filtering logic. r=abr 2014-01-09 15:12:25 -08:00
Byron Campen [:bwc]
a14dd974ac Bug 786234 - Part 2: Implementation of the filtering logic itself, plus a unit-test. r=abr 2013-12-19 16:19:05 -08:00
Gian-Carlo Pascutto
4848b0287e Bug 978827 - Remove bogus assertion in OpenSL input backend. r=jesup 2014-03-06 15:11:52 +01:00
Byron Campen [:bwc]
bc2949dab7 Bug 970690 - Part 1: Add transport field to ICE candidate stats. r=jib 2014-02-26 11:12:37 -08:00
Byron Campen [:bwc]
88008dfd1d Bug 958221 - Part 3: New webidl for WebrtcGlobalInformation, c++ impl, and removing logging-related stuff from PeerConnectionImpl. r=jib 2014-02-20 09:35:35 -08:00
Byron Campen [:bwc]
6322dddd87 Bug 958221 - Part 2: Simplify the statistics accessors, and rename things to better describe what they do. r=jib 2014-02-20 09:33:55 -08:00
Byron Campen [:bwc]
0044ef7d1c Bug 958221 - Part 1: Move (but not alter) some code as step one in refactoring the GetStats functionality so it can be used by other classes. r=jib 2014-01-24 14:48:15 -08:00
Ulrich Weigand
5d45690169 Bug 976648 - WebRTC endian config for powerpc64le-linux support. r=jesup 2014-02-28 09:57:03 -05:00
Ryan VanderMeulen
760e2a311f Backed out changeset 6f05267b4afc (bug 798033) for Android bustage. 2014-03-05 09:55:52 -05:00
snigdha
ef5d7c4e73 Bug 798033 - Headers should generally not do "using namespace" at file scope. r=jib, r=jmathies, r=rjesup, r=ekr, r=ncameron, r=blassey 2014-03-05 08:47:45 -05:00
Jan-Ivar Bruaroey
8576953acf Bug 978239 - Synchronize WindowsRealTimeClock to unmess RTCP timestamps. r=jesup 2014-02-28 15:42:24 -05:00
Gian-Carlo Pascutto
784c8ca252 Bug 974378 - Make webrtc.org OpenSL ES output code optional. Increase input buffers. r=jesup 2014-02-26 19:55:07 +01:00
Martin Thomson
6a3f07e86b Bug 884573 - Part 2: Identity assertion generation and verification for WebRTC. r=abr 2014-02-10 14:41:46 -08:00
Martin Thomson
195ba6315f Bug 884573 - Part 1: Add a=identity support for sipcc. r=abr 2014-01-08 11:55:04 -08:00